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Highlighted
Beginner

Block incoming anonymous calls

Hi,

I'm trying to block incoming blocked/unknown/anonymous callers over a sip trunk? I've creaed a translation rule and applied it:

voice translation-rule 5000

rule 1 reject /^$/

voice translation-profile CallBlock5000

translate calling 5000

dial-peer voice xxxx voip

call-block translation-profile incoming CallBlock5000

call-block disconnect-cause incoming invalid-number

To try it out, I'm dialing (from a normal/off network cell and landline) *67 and then the number. This does not work; only if I match the exact number I'm calling from, then it does get blocked.

When I show sip calls during the *67 call I see the calling number is blank.

   Calling Number         :

When I show sip calls during the regular call, I see the proper Calling Number.

As I understand it, with Call Manager and phones running SCCP, I cannot enable/use anonymous call blocking; so I do have to enforce the call blocking policy at this gateway device (UC520).

I'm very new to Cisco voice, so sorry I'f I'm missing something obvious. Thanks in advance!

11 REPLIES 11
Highlighted
Rising star

What is the anonymization method used by your isp provider?

In SIP you can set CLIR in various ways:

- set the display info of from header to anonymous but not the user part;

- set the user part of from to anonymous but not domain;

- use the form anonymous@anonymous.invalid in the from header;

- set the privacy option in Remote Party ID;

- set the option Privacy in P-Asserter-ID;

- remove identity omitting the number;

etc.

Can you add the output of debug ccsip messages?

In this way we can find what methods are used.

In the meantime you can test this rule:

rule 1 reject /^.*/ type reserved

Regards.

Highlighted

Thanks, I tried the rule--still private gets through.

*Feb 14 21:52:37.457: //394/0B18CF2B8382/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 207.2.123.180:5060;branch=z9hG4bK71b5401f596084541bb0894ae16bbbc8

From: <6103506982>;tag=3538245046-512161

To: <>3023522325@aa.aa.aa.aa(CUBE external IP)>

Date: Tue, 14 Feb 2012 21:52:37 GMT

Call-ID: 461523831-3538245046-512158@msc7.mydomain.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

*Feb 14 21:52:37.457: //395/0B18CF2B8382/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:3023522325@xx.xx.xx.xx(UCMBE IP):5060 SIP/2.0

Via: SIP/2.0/UDP yy.yy.yy.yy(UC500/CUBE IP)>;:5060;branch=z9hG4bK2B4F4E

Remote-Party-ID: <>6103506982@yy.yy.yy.yy(UC500/CUBE IP)>;party=calling;screen=no;privacy=full

From: "anonymous" ;tag=4C78DEC-1E4B

To: <>3023522325@xx.xx.xx.xx>

Date: Tue, 14 Feb 2012 21:52:37 GMT

Call-ID: B1B3FCA-568D11E1-83889F30-EB9D8BD5@192.168.20.5

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE: 1800

Cisco-Guid: 0186175275-1452085729-2206375728-3952970709

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Timestamp: 1329256357

Contact:

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 69

Session-Expires: 3600;refresher=uac

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 246

v=0

o=CiscoSystemsSIP-GW-UserAgent 2063 472 IN IP4 yy.yy.yy.yy(UC500/CUBE IP)

s=SIP Call

c=IN IP4 yy.yy.yy.yy(UC500/CUBE IP)

t=0 0

m=audio 19566 RTP/AVP 0 101

c=IN IP4 192.168.20.5

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

*Feb 14 21:52:37.465: //395/0B18CF2B8382/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Date: Tue, 14 Feb 2012 21:50:47 GMT

From: "anonymous" ;tag=4C78DEC-1E4B

Allow-Events: presence

Content-Length: 0

To: <>3023522325@xx.xx.xx.xx(UCMBE IP)>

Call-ID: B1B3FCA-568D11E1-83889F30-EB9D8BD5@192.168.20.5

Via: SIP/2.0/UDP yy.yy.yy.yy(UC500/CUBE IP):5060;branch=z9hG4bK2B4F4E

CSeq: 101 INVITE

*Feb 14 21:52:37.469: //395/0B18CF2B8382/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 180 Ringing

Date: Tue, 14 Feb 2012 21:50:47 GMT

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY

From: "anonymous" ;tag=4C78DEC-1E4B

Highlighted

So, I tried tried the rule /^.*/, and adding a rule to the sip profile to convert the anonymous@ip to anonymous@anonymou.invalid, but the calls still get through. Any comment on the debug output? Thanks

Highlighted

Try a SIP profile, eliminating Remote-party-ID altogether.

Highlighted
Beginner

Try the following:

voice translation-rule 5000

rule 1 reject /^$/

rule 2 reject /[^0-9]/

Highlighted

Have the same request and followed along but also not having success with getting it to work. 

rule 1 reject /^$/  - Does not seem to match the unknown/private/anonymous calls

rule 2 reject /[^0-9]/ - Matches, however once the rule is put in any/all calls are rejected/blocked 

Also, if a rule is added with a specific number, then that number will match and the call is rejected. 

Thanks for any additional info!

Highlighted

The command "rule 2 reject /[^0-9]/" matches any CID that contains anything but a number.  A similar command would be "rule 2 reject /^[^0-9]/", which means match CID that does not begin with a number.  Try that.

For SIP calls, when a call is passed with no calling party number, the calling party number is set to anonymous, which does not match the "rule 1 reject /^$/" command because the calling party number is not null.  Check the debugs to see exactly what the calling party number is.

Highlighted

Thanks for the super quick response.

rule 2 reject /^[^0-9]/  also appears to match any/all calls, but looking at the SIP debugs, it would seem that's happening because the SIP provider is sending inbound calls with a + mark before the number. 

Here's an example of a non private call:

From: "CID-Name" <>

An example of a private/anonymous call (*67 before dialing):

From: "Anonymous"

Any possible suggestions to work around that?   Thanks! 

Highlighted

Try "rule 2 reject /[^0-9]$/".

Here is a good link to read up on regarding translation rules:

http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml

Highlighted

That worked, thanks!!

Highlighted
Beginner

Also, try modifying the disconnect cause to call-reject if the above doesn't work.

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